2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
21 * @short_description: A media stream
22 * @see_also: #GstRTSPMedia
24 * The #GstRTSPStream object manages the data transport for one stream. It
25 * is created from a payloader element and a source pad that produce the RTP
26 * packets for the stream.
28 * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
29 * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
31 * The #GstRTSPStream will use the configured addresspool, as set with
32 * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
33 * stream. With gst_rtsp_stream_get_multicast_address() you can get the
36 * With gst_rtsp_stream_get_server_port () you can get the port that the server
37 * will use to receive RTCP. This is the part that the clients will use to send
40 * With gst_rtsp_stream_add_transport() destinations can be added where the
41 * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
42 * the destination again.
44 * Last reviewed on 2013-07-16 (1.0.0)
53 #include <gst/app/gstappsrc.h>
54 #include <gst/app/gstappsink.h>
56 #include "rtsp-stream.h"
58 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
59 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
61 struct _GstRTSPStreamPrivate
66 GstElement *payloader;
71 GstRTSPLowerTrans protocols;
73 /* pads on the rtpbin */
74 GstPad *send_rtp_sink;
78 /* the RTPSession object */
81 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
83 GstElement *udpsrc_v4[2];
85 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
87 GstElement *udpsrc_v6[2];
89 GstElement *udpsink[2];
91 /* for TCP transport */
92 GstElement *appsrc[2];
93 GstElement *appqueue[2];
94 GstElement *appsink[2];
97 GstElement *funnel[2];
99 /* server ports for sending/receiving over ipv4 */
100 GstRTSPRange server_port_v4;
101 GstRTSPAddress *server_addr_v4;
104 /* server ports for sending/receiving over ipv6 */
105 GstRTSPRange server_port_v6;
106 GstRTSPAddress *server_addr_v6;
109 /* multicast addresses */
110 GstRTSPAddressPool *pool;
111 GstRTSPAddress *addr_v4;
112 GstRTSPAddress *addr_v6;
114 /* the caps of the stream */
118 /* transports we stream to */
125 #define DEFAULT_CONTROL NULL
126 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
127 GST_RTSP_LOWER_TRANS_TCP
137 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
138 #define GST_CAT_DEFAULT rtsp_stream_debug
140 static GQuark ssrc_stream_map_key;
142 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
143 GValue * value, GParamSpec * pspec);
144 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
145 const GValue * value, GParamSpec * pspec);
147 static void gst_rtsp_stream_finalize (GObject * obj);
149 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
152 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
154 GObjectClass *gobject_class;
156 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
158 gobject_class = G_OBJECT_CLASS (klass);
160 gobject_class->get_property = gst_rtsp_stream_get_property;
161 gobject_class->set_property = gst_rtsp_stream_set_property;
162 gobject_class->finalize = gst_rtsp_stream_finalize;
164 g_object_class_install_property (gobject_class, PROP_CONTROL,
165 g_param_spec_string ("control", "Control",
166 "The control string for this stream", DEFAULT_CONTROL,
167 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
169 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
170 g_param_spec_flags ("protocols", "Protocols",
171 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
172 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
174 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
176 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
180 gst_rtsp_stream_init (GstRTSPStream * stream)
182 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
184 GST_DEBUG ("new stream %p", stream);
189 priv->control = g_strdup (DEFAULT_CONTROL);
190 priv->protocols = DEFAULT_PROTOCOLS;
192 g_mutex_init (&priv->lock);
196 gst_rtsp_stream_finalize (GObject * obj)
198 GstRTSPStream *stream;
199 GstRTSPStreamPrivate *priv;
201 stream = GST_RTSP_STREAM (obj);
204 GST_DEBUG ("finalize stream %p", stream);
206 /* we really need to be unjoined now */
207 g_return_if_fail (!priv->is_joined);
210 gst_rtsp_address_free (priv->addr_v4);
212 gst_rtsp_address_free (priv->addr_v6);
213 if (priv->server_addr_v4)
214 gst_rtsp_address_free (priv->server_addr_v4);
215 if (priv->server_addr_v6)
216 gst_rtsp_address_free (priv->server_addr_v6);
218 g_object_unref (priv->pool);
219 gst_object_unref (priv->payloader);
220 gst_object_unref (priv->srcpad);
221 g_free (priv->control);
222 g_mutex_clear (&priv->lock);
224 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
228 gst_rtsp_stream_get_property (GObject * object, guint propid,
229 GValue * value, GParamSpec * pspec)
231 GstRTSPStream *stream = GST_RTSP_STREAM (object);
235 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
238 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
241 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
246 gst_rtsp_stream_set_property (GObject * object, guint propid,
247 const GValue * value, GParamSpec * pspec)
249 GstRTSPStream *stream = GST_RTSP_STREAM (object);
253 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
256 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
259 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
264 * gst_rtsp_stream_new:
267 * @payloader: a #GstElement
269 * Create a new media stream with index @idx that handles RTP data on
270 * @srcpad and has a payloader element @payloader.
272 * Returns: a new #GstRTSPStream
275 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
277 GstRTSPStreamPrivate *priv;
278 GstRTSPStream *stream;
280 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
281 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
282 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
284 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
287 priv->payloader = gst_object_ref (payloader);
288 priv->srcpad = gst_object_ref (srcpad);
294 * gst_rtsp_stream_get_index:
295 * @stream: a #GstRTSPStream
297 * Get the stream index.
299 * Return: the stream index.
302 gst_rtsp_stream_get_index (GstRTSPStream * stream)
304 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
306 return stream->priv->idx;
310 * gst_rtsp_stream_get_srcpad:
311 * @stream: a #GstRTSPStream
313 * Get the srcpad associated with @stream.
315 * Return: the srcpad. Unref after usage.
318 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
320 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
322 return gst_object_ref (stream->priv->srcpad);
326 * gst_rtsp_stream_get_control:
327 * @stream: a #GstRTSPStream
329 * Get the control string to identify this stream.
331 * Return: the control string. free after usage.
334 gst_rtsp_stream_get_control (GstRTSPStream * stream)
336 GstRTSPStreamPrivate *priv;
339 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
343 g_mutex_lock (&priv->lock);
344 if ((result = g_strdup (priv->control)) == NULL)
345 result = g_strdup_printf ("stream=%u", priv->idx);
346 g_mutex_unlock (&priv->lock);
352 * gst_rtsp_stream_set_control:
353 * @stream: a #GstRTSPStream
354 * @control: a control string
356 * Set the control string in @stream.
359 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
361 GstRTSPStreamPrivate *priv;
363 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
367 g_mutex_lock (&priv->lock);
368 g_free (priv->control);
369 priv->control = g_strdup (control);
370 g_mutex_unlock (&priv->lock);
374 * gst_rtsp_stream_has_control:
375 * @stream: a #GstRTSPStream
376 * @control: a control string
378 * Check if @stream has the control string @control.
380 * Returns: %TRUE is @stream has @control as the control string
383 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
385 GstRTSPStreamPrivate *priv;
388 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
392 g_mutex_lock (&priv->lock);
394 res = g_strcmp0 (priv->control, control);
397 sscanf (control, "stream=%u", &streamid);
398 res = (streamid == priv->idx);
400 g_mutex_unlock (&priv->lock);
406 * gst_rtsp_stream_set_mtu:
407 * @stream: a #GstRTSPStream
410 * Configure the mtu in the payloader of @stream to @mtu.
413 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
415 GstRTSPStreamPrivate *priv;
417 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
421 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
423 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
427 * gst_rtsp_stream_get_mtu:
428 * @stream: a #GstRTSPStream
430 * Get the configured MTU in the payloader of @stream.
432 * Returns: the MTU of the payloader.
435 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
437 GstRTSPStreamPrivate *priv;
440 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
444 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
449 /* Update the dscp qos property on the udp sinks */
451 update_dscp_qos (GstRTSPStream * stream)
453 GstRTSPStreamPrivate *priv;
455 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
459 if (priv->udpsink[0]) {
460 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
464 if (priv->udpsink[1]) {
465 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
471 * gst_rtsp_stream_set_dscp_qos:
472 * @stream: a #GstRTSPStream
473 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
475 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
478 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
480 GstRTSPStreamPrivate *priv;
482 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
486 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
488 if (dscp_qos < -1 || dscp_qos > 63) {
489 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
493 priv->dscp_qos = dscp_qos;
495 update_dscp_qos (stream);
499 * gst_rtsp_stream_get_dscp_qos:
500 * @stream: a #GstRTSPStream
502 * Get the configured DSCP QoS in of the outgoing sockets.
504 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
507 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
509 GstRTSPStreamPrivate *priv;
511 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
515 return priv->dscp_qos;
519 * gst_rtsp_stream_set_protocols:
520 * @stream: a #GstRTSPStream
521 * @protocols: the new flags
523 * Configure the allowed lower transport for @stream.
526 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
527 GstRTSPLowerTrans protocols)
529 GstRTSPStreamPrivate *priv;
531 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
535 g_mutex_lock (&priv->lock);
536 priv->protocols = protocols;
537 g_mutex_unlock (&priv->lock);
541 * gst_rtsp_stream_get_protocols:
542 * @stream: a #GstRTSPStream
544 * Get the allowed protocols of @stream.
546 * Returns: a #GstRTSPLowerTrans
549 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
551 GstRTSPStreamPrivate *priv;
552 GstRTSPLowerTrans res;
554 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
555 GST_RTSP_LOWER_TRANS_UNKNOWN);
559 g_mutex_lock (&priv->lock);
560 res = priv->protocols;
561 g_mutex_unlock (&priv->lock);
567 * gst_rtsp_stream_set_address_pool:
568 * @stream: a #GstRTSPStream
569 * @pool: a #GstRTSPAddressPool
571 * configure @pool to be used as the address pool of @stream.
574 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
575 GstRTSPAddressPool * pool)
577 GstRTSPStreamPrivate *priv;
578 GstRTSPAddressPool *old;
580 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
584 GST_LOG_OBJECT (stream, "set address pool %p", pool);
586 g_mutex_lock (&priv->lock);
587 if ((old = priv->pool) != pool)
588 priv->pool = pool ? g_object_ref (pool) : NULL;
591 g_mutex_unlock (&priv->lock);
594 g_object_unref (old);
598 * gst_rtsp_stream_get_address_pool:
599 * @stream: a #GstRTSPStream
601 * Get the #GstRTSPAddressPool used as the address pool of @stream.
603 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
607 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
609 GstRTSPStreamPrivate *priv;
610 GstRTSPAddressPool *result;
612 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
616 g_mutex_lock (&priv->lock);
617 if ((result = priv->pool))
618 g_object_ref (result);
619 g_mutex_unlock (&priv->lock);
625 * gst_rtsp_stream_get_multicast_address:
626 * @stream: a #GstRTSPStream
627 * @family: the #GSocketFamily
629 * Get the multicast address of @stream for @family.
631 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
632 * allocated. gst_rtsp_address_free() after usage.
635 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
636 GSocketFamily family)
638 GstRTSPStreamPrivate *priv;
639 GstRTSPAddress *result;
640 GstRTSPAddress **addrp;
641 GstRTSPAddressFlags flags;
643 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
647 if (family == G_SOCKET_FAMILY_IPV6) {
648 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
649 addrp = &priv->addr_v4;
651 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
652 addrp = &priv->addr_v6;
655 g_mutex_lock (&priv->lock);
656 if (*addrp == NULL) {
657 if (priv->pool == NULL)
660 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
662 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
666 result = gst_rtsp_address_copy (*addrp);
667 g_mutex_unlock (&priv->lock);
674 GST_ERROR_OBJECT (stream, "no address pool specified");
675 g_mutex_unlock (&priv->lock);
680 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
681 g_mutex_unlock (&priv->lock);
687 * gst_rtsp_stream_reserve_address:
688 * @stream: a #GstRTSPStream
689 * @address: an address
694 * Reserve @address and @port as the address and port of @stream.
696 * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
697 * reserved. gst_rtsp_address_free() after usage.
700 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
701 const gchar * address, guint port, guint n_ports, guint ttl)
703 GstRTSPStreamPrivate *priv;
704 GstRTSPAddress *result;
706 GSocketFamily family;
707 GstRTSPAddress **addrp;
709 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
710 g_return_val_if_fail (address != NULL, NULL);
711 g_return_val_if_fail (port > 0, NULL);
712 g_return_val_if_fail (n_ports > 0, NULL);
713 g_return_val_if_fail (ttl > 0, NULL);
717 addr = g_inet_address_new_from_string (address);
719 GST_ERROR ("failed to get inet addr from %s", address);
720 family = G_SOCKET_FAMILY_IPV4;
722 family = g_inet_address_get_family (addr);
723 g_object_unref (addr);
726 if (family == G_SOCKET_FAMILY_IPV6)
727 addrp = &priv->addr_v4;
729 addrp = &priv->addr_v6;
731 g_mutex_lock (&priv->lock);
732 if (*addrp == NULL) {
733 if (priv->pool == NULL)
736 *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
741 if (strcmp ((*addrp)->address, address) ||
742 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
743 (*addrp)->ttl != ttl)
744 goto different_address;
746 result = gst_rtsp_address_copy (*addrp);
747 g_mutex_unlock (&priv->lock);
754 GST_ERROR_OBJECT (stream, "no address pool specified");
755 g_mutex_unlock (&priv->lock);
760 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
762 g_mutex_unlock (&priv->lock);
767 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
768 " reserved", address);
769 g_mutex_unlock (&priv->lock);
775 alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
776 GSocketFamily family, GstElement * udpsrc_out[2],
777 GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
778 GstRTSPAddress ** server_addr_out)
780 GstStateChangeReturn ret;
781 GstElement *udpsrc0, *udpsrc1;
782 GstElement *udpsink0, *udpsink1;
783 GSocket *rtp_socket = NULL;
784 GSocket *rtcp_socket;
785 gint tmp_rtp, tmp_rtcp;
787 gint rtpport, rtcpport;
788 GList *rejected_addresses = NULL;
789 GstRTSPAddress *addr = NULL;
790 GInetAddress *inetaddr = NULL;
791 GSocketAddress *rtp_sockaddr = NULL;
792 GSocketAddress *rtcp_sockaddr = NULL;
793 const gchar *multisink_socket;
795 if (family == G_SOCKET_FAMILY_IPV6)
796 multisink_socket = "socket-v6";
798 multisink_socket = "socket";
806 /* Start with random port */
809 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
810 G_SOCKET_PROTOCOL_UDP, NULL);
812 goto no_udp_protocol;
814 if (*server_addr_out)
815 gst_rtsp_address_free (*server_addr_out);
817 /* try to allocate 2 UDP ports, the RTP port should be an even
818 * number and the RTCP port should be the next (uneven) port */
821 if (rtp_socket == NULL) {
822 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
823 G_SOCKET_PROTOCOL_UDP, NULL);
825 goto no_udp_protocol;
828 if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
829 GstRTSPAddressFlags flags;
832 rejected_addresses = g_list_prepend (rejected_addresses, addr);
834 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
835 if (family == G_SOCKET_FAMILY_IPV6)
836 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
838 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
840 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
845 tmp_rtp = addr->port;
847 g_clear_object (&inetaddr);
848 inetaddr = g_inet_address_new_from_string (addr->address);
856 if (inetaddr == NULL)
857 inetaddr = g_inet_address_new_any (family);
860 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
861 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
862 g_object_unref (rtp_sockaddr);
865 g_object_unref (rtp_sockaddr);
867 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
868 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
869 g_clear_object (&rtp_sockaddr);
874 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
875 g_object_unref (rtp_sockaddr);
877 /* check if port is even */
878 if ((tmp_rtp & 1) != 0) {
879 /* port not even, close and allocate another */
881 g_clear_object (&rtp_socket);
886 tmp_rtcp = tmp_rtp + 1;
888 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
889 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
890 g_object_unref (rtcp_sockaddr);
891 g_clear_object (&rtp_socket);
894 g_object_unref (rtcp_sockaddr);
896 g_clear_object (&inetaddr);
898 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
899 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
901 if (udpsrc0 == NULL || udpsrc1 == NULL)
902 goto no_udp_protocol;
904 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
905 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
907 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
908 if (ret == GST_STATE_CHANGE_FAILURE)
910 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
911 if (ret == GST_STATE_CHANGE_FAILURE)
914 /* all fine, do port check */
915 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
916 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
918 /* this should not happen... */
919 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
923 udpsink0 = udpsink_out[0];
925 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
928 goto no_udp_protocol;
930 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
931 g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
934 udpsink1 = udpsink_out[1];
936 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
939 goto no_udp_protocol;
941 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
942 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
943 g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
945 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
946 g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
947 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
948 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
949 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
950 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
951 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
952 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
954 /* we keep these elements, we will further configure them when the
955 * client told us to really use the UDP ports. */
956 udpsrc_out[0] = udpsrc0;
957 udpsrc_out[1] = udpsrc1;
958 udpsink_out[0] = udpsink0;
959 udpsink_out[1] = udpsink1;
960 server_port_out->min = rtpport;
961 server_port_out->max = rtcpport;
963 *server_addr_out = addr;
964 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
966 g_object_unref (rtp_socket);
967 g_object_unref (rtcp_socket);
995 gst_element_set_state (udpsrc0, GST_STATE_NULL);
996 gst_object_unref (udpsrc0);
999 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1000 gst_object_unref (udpsrc1);
1003 gst_element_set_state (udpsink0, GST_STATE_NULL);
1004 gst_object_unref (udpsink0);
1007 gst_element_set_state (udpsink1, GST_STATE_NULL);
1008 gst_object_unref (udpsink1);
1011 g_object_unref (inetaddr);
1012 g_list_free_full (rejected_addresses,
1013 (GDestroyNotify) gst_rtsp_address_free);
1015 gst_rtsp_address_free (addr);
1017 g_object_unref (rtp_socket);
1019 g_object_unref (rtcp_socket);
1024 /* must be called with lock */
1026 alloc_ports (GstRTSPStream * stream)
1028 GstRTSPStreamPrivate *priv = stream->priv;
1030 priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1031 G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
1032 &priv->server_port_v4, &priv->server_addr_v4);
1034 priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
1035 G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
1036 &priv->server_port_v6, &priv->server_addr_v6);
1038 return priv->have_ipv4 || priv->have_ipv6;
1042 * gst_rtsp_stream_get_server_port:
1043 * @stream: a #GstRTSPStream
1044 * @server_port: (out): result server port
1045 * @family: the port family to get
1047 * Fill @server_port with the port pair used by the server. This function can
1048 * only be called when @stream has been joined.
1051 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1052 GstRTSPRange * server_port, GSocketFamily family)
1054 GstRTSPStreamPrivate *priv;
1056 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1057 priv = stream->priv;
1058 g_return_if_fail (priv->is_joined);
1060 g_mutex_lock (&priv->lock);
1061 if (family == G_SOCKET_FAMILY_IPV4) {
1063 *server_port = priv->server_port_v4;
1066 *server_port = priv->server_port_v6;
1068 g_mutex_unlock (&priv->lock);
1072 * gst_rtsp_stream_get_rtpsession:
1073 * @stream: a #GstRTSPStream
1075 * Get the RTP session of this stream.
1077 * Returns: The RTP session of this stream. Unref after usage.
1080 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1082 GstRTSPStreamPrivate *priv;
1085 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1087 priv = stream->priv;
1089 g_mutex_lock (&priv->lock);
1090 if ((session = priv->session))
1091 g_object_ref (session);
1092 g_mutex_unlock (&priv->lock);
1098 * gst_rtsp_stream_get_ssrc:
1099 * @stream: a #GstRTSPStream
1100 * @ssrc: (out): result ssrc
1102 * Get the SSRC used by the RTP session of this stream. This function can only
1103 * be called when @stream has been joined.
1106 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1108 GstRTSPStreamPrivate *priv;
1110 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1111 priv = stream->priv;
1112 g_return_if_fail (priv->is_joined);
1114 g_mutex_lock (&priv->lock);
1115 if (ssrc && priv->session)
1116 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1117 g_mutex_unlock (&priv->lock);
1120 /* executed from streaming thread */
1122 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1124 GstRTSPStreamPrivate *priv = stream->priv;
1125 GstCaps *newcaps, *oldcaps;
1127 newcaps = gst_pad_get_current_caps (pad);
1129 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1132 g_mutex_lock (&priv->lock);
1133 oldcaps = priv->caps;
1134 priv->caps = newcaps;
1135 g_mutex_unlock (&priv->lock);
1138 gst_caps_unref (oldcaps);
1142 dump_structure (const GstStructure * s)
1146 sstr = gst_structure_to_string (s);
1147 GST_INFO ("structure: %s", sstr);
1151 static GstRTSPStreamTransport *
1152 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1154 GstRTSPStreamPrivate *priv = stream->priv;
1156 GstRTSPStreamTransport *result = NULL;
1161 if (rtcp_from == NULL)
1164 tmp = g_strrstr (rtcp_from, ":");
1168 port = atoi (tmp + 1);
1169 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1171 g_mutex_lock (&priv->lock);
1172 GST_INFO ("finding %s:%d in %d transports", dest, port,
1173 g_list_length (priv->transports));
1175 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1176 GstRTSPStreamTransport *trans = walk->data;
1177 const GstRTSPTransport *tr;
1180 tr = gst_rtsp_stream_transport_get_transport (trans);
1182 min = tr->client_port.min;
1183 max = tr->client_port.max;
1185 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1191 g_object_ref (result);
1192 g_mutex_unlock (&priv->lock);
1199 static GstRTSPStreamTransport *
1200 check_transport (GObject * source, GstRTSPStream * stream)
1202 GstStructure *stats;
1203 GstRTSPStreamTransport *trans;
1205 /* see if we have a stream to match with the origin of the RTCP packet */
1206 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1207 if (trans == NULL) {
1208 g_object_get (source, "stats", &stats, NULL);
1210 const gchar *rtcp_from;
1212 dump_structure (stats);
1214 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1215 if ((trans = find_transport (stream, rtcp_from))) {
1216 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1218 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1221 gst_structure_free (stats);
1229 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1231 GstRTSPStreamTransport *trans;
1233 GST_INFO ("%p: new source %p", stream, source);
1235 trans = check_transport (source, stream);
1238 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1242 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1244 GST_INFO ("%p: new SDES %p", stream, source);
1248 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1250 GstRTSPStreamTransport *trans;
1252 trans = check_transport (source, stream);
1255 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1256 gst_rtsp_stream_transport_keep_alive (trans);
1260 GstStructure *stats;
1261 g_object_get (source, "stats", &stats, NULL);
1263 dump_structure (stats);
1264 gst_structure_free (stats);
1271 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1273 GST_INFO ("%p: source %p bye", stream, source);
1277 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1279 GstRTSPStreamTransport *trans;
1281 GST_INFO ("%p: source %p bye timeout", stream, source);
1283 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1284 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1285 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1290 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1292 GstRTSPStreamTransport *trans;
1294 GST_INFO ("%p: source %p timeout", stream, source);
1296 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1297 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1298 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1302 static GstFlowReturn
1303 handle_new_sample (GstAppSink * sink, gpointer user_data)
1305 GstRTSPStreamPrivate *priv;
1309 GstRTSPStream *stream;
1311 sample = gst_app_sink_pull_sample (sink);
1315 stream = (GstRTSPStream *) user_data;
1316 priv = stream->priv;
1317 buffer = gst_sample_get_buffer (sample);
1319 g_mutex_lock (&priv->lock);
1320 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1321 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1323 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
1324 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1326 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
1329 g_mutex_unlock (&priv->lock);
1331 gst_sample_unref (sample);
1336 static GstAppSinkCallbacks sink_cb = {
1337 NULL, /* not interested in EOS */
1338 NULL, /* not interested in preroll samples */
1343 * gst_rtsp_stream_join_bin:
1344 * @stream: a #GstRTSPStream
1345 * @bin: a #GstBin to join
1346 * @rtpbin: a rtpbin element in @bin
1347 * @state: the target state of the new elements
1349 * Join the #Gstbin @bin that contains the element @rtpbin.
1351 * @stream will link to @rtpbin, which must be inside @bin. The elements
1352 * added to @bin will be set to the state given in @state.
1354 * Returns: %TRUE on success.
1357 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
1358 GstElement * rtpbin, GstState state)
1360 GstRTSPStreamPrivate *priv;
1364 GstPad *pad, *sinkpad, *selpad;
1365 GstPadLinkReturn ret;
1367 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1368 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1369 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1371 priv = stream->priv;
1373 g_mutex_lock (&priv->lock);
1374 if (priv->is_joined)
1377 /* create a session with the same index as the stream */
1380 GST_INFO ("stream %p joining bin as session %u", stream, idx);
1382 if (!alloc_ports (stream))
1385 /* update the dscp qos field in the sinks */
1386 update_dscp_qos (stream);
1388 /* get a pad for sending RTP */
1389 name = g_strdup_printf ("send_rtp_sink_%u", idx);
1390 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
1392 /* link the RTP pad to the session manager, it should not really fail unless
1393 * this is not really an RTP pad */
1394 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
1395 if (ret != GST_PAD_LINK_OK)
1398 /* get pads from the RTP session element for sending and receiving
1400 name = g_strdup_printf ("send_rtp_src_%u", idx);
1401 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
1403 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1404 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1406 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1407 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1409 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1410 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1413 /* get the session */
1414 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1416 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1418 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1420 g_signal_connect (priv->session, "on-ssrc-active",
1421 (GCallback) on_ssrc_active, stream);
1422 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1424 g_signal_connect (priv->session, "on-bye-timeout",
1425 (GCallback) on_bye_timeout, stream);
1426 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1429 for (i = 0; i < 2; i++) {
1430 GstPad *teepad, *queuepad;
1431 /* For the sender we create this bit of pipeline for both
1432 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1433 * we need to add a queue before appsink to make the pipeline
1434 * not block. For the TCP case, we want to pump data to the
1435 * client as fast as possible anyway.
1437 * .--------. .-----. .---------.
1438 * | rtpbin | | tee | | udpsink |
1439 * | send->sink src->sink |
1440 * '--------' | | '---------'
1441 * | | .---------. .---------.
1442 * | | | queue | | appsink |
1443 * | src->sink src->sink |
1444 * '-----' '---------' '---------'
1446 * When only UDP is allowed, we skip the tee, queue and appsink and link the
1447 * udpsink directly to the session.
1450 gst_bin_add (bin, priv->udpsink[i]);
1451 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1453 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1454 /* make tee for RTP/RTCP */
1455 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1456 gst_bin_add (bin, priv->tee[i]);
1458 /* and link to rtpbin send pad */
1459 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1460 gst_pad_link (priv->send_src[i], pad);
1461 gst_object_unref (pad);
1463 /* link tee to udpsink */
1464 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1465 gst_pad_link (teepad, sinkpad);
1466 gst_object_unref (teepad);
1469 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1470 gst_bin_add (bin, priv->appqueue[i]);
1471 /* and link to tee */
1472 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1473 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1474 gst_pad_link (teepad, pad);
1475 gst_object_unref (pad);
1476 gst_object_unref (teepad);
1479 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1480 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1481 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1482 gst_bin_add (bin, priv->appsink[i]);
1483 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1484 &sink_cb, stream, NULL);
1485 /* and link to queue */
1486 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1487 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1488 gst_pad_link (queuepad, pad);
1489 gst_object_unref (pad);
1490 gst_object_unref (queuepad);
1492 /* else only udpsink needed, link it to the session */
1493 gst_pad_link (priv->send_src[i], sinkpad);
1495 gst_object_unref (sinkpad);
1497 /* For the receiver we create this bit of pipeline for both
1498 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1499 * and it is all funneled into the rtpbin receive pad.
1501 * .--------. .--------. .--------.
1502 * | udpsrc | | funnel | | rtpbin |
1503 * | src->sink src->sink |
1504 * '--------' | | '--------'
1508 * '--------' '--------'
1510 /* make funnel for the RTP/RTCP receivers */
1511 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1512 gst_bin_add (bin, priv->funnel[i]);
1514 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1515 gst_pad_link (pad, priv->recv_sink[i]);
1516 gst_object_unref (pad);
1518 if (priv->udpsrc_v4[i]) {
1519 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1521 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
1522 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
1524 gst_bin_add (bin, priv->udpsrc_v4[i]);
1526 /* and link to the funnel v4 */
1527 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1528 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
1529 gst_pad_link (pad, selpad);
1530 gst_object_unref (pad);
1531 gst_object_unref (selpad);
1534 if (priv->udpsrc_v6[i]) {
1535 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
1536 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
1537 gst_bin_add (bin, priv->udpsrc_v6[i]);
1539 /* and link to the funnel v6 */
1540 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1541 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
1542 gst_pad_link (pad, selpad);
1543 gst_object_unref (pad);
1544 gst_object_unref (selpad);
1547 if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
1548 /* make and add appsrc */
1549 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1550 gst_bin_add (bin, priv->appsrc[i]);
1551 /* and link to the funnel */
1552 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1553 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1554 gst_pad_link (pad, selpad);
1555 gst_object_unref (pad);
1556 gst_object_unref (selpad);
1559 /* check if we need to set to a special state */
1560 if (state != GST_STATE_NULL) {
1561 if (priv->udpsink[i])
1562 gst_element_set_state (priv->udpsink[i], state);
1563 if (priv->appsink[i])
1564 gst_element_set_state (priv->appsink[i], state);
1565 if (priv->appqueue[i])
1566 gst_element_set_state (priv->appqueue[i], state);
1568 gst_element_set_state (priv->tee[i], state);
1569 if (priv->funnel[i])
1570 gst_element_set_state (priv->funnel[i], state);
1571 if (priv->appsrc[i])
1572 gst_element_set_state (priv->appsrc[i], state);
1576 /* be notified of caps changes */
1577 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1578 (GCallback) caps_notify, stream);
1580 priv->is_joined = TRUE;
1581 g_mutex_unlock (&priv->lock);
1588 g_mutex_unlock (&priv->lock);
1593 g_mutex_unlock (&priv->lock);
1594 GST_WARNING ("failed to allocate ports %u", idx);
1599 GST_WARNING ("failed to link stream %u", idx);
1600 gst_object_unref (priv->send_rtp_sink);
1601 priv->send_rtp_sink = NULL;
1602 g_mutex_unlock (&priv->lock);
1608 * gst_rtsp_stream_leave_bin:
1609 * @stream: a #GstRTSPStream
1611 * @rtpbin: a rtpbin #GstElement
1613 * Remove the elements of @stream from @bin.
1615 * Return: %TRUE on success.
1618 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1619 GstElement * rtpbin)
1621 GstRTSPStreamPrivate *priv;
1624 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1625 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1626 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1628 priv = stream->priv;
1630 g_mutex_lock (&priv->lock);
1631 if (!priv->is_joined)
1632 goto was_not_joined;
1634 /* all transports must be removed by now */
1635 g_return_val_if_fail (priv->transports == NULL, FALSE);
1637 GST_INFO ("stream %p leaving bin", stream);
1639 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1640 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1641 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1642 gst_object_unref (priv->send_rtp_sink);
1643 priv->send_rtp_sink = NULL;
1645 for (i = 0; i < 2; i++) {
1646 if (priv->udpsink[i])
1647 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1648 if (priv->appsink[i])
1649 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1650 if (priv->appqueue[i])
1651 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1653 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1654 if (priv->funnel[i])
1655 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1656 if (priv->appsrc[i])
1657 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1658 if (priv->udpsrc_v4[i]) {
1659 /* and set udpsrc to NULL now before removing */
1660 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
1661 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
1662 /* removing them should also nicely release the request
1663 * pads when they finalize */
1664 gst_bin_remove (bin, priv->udpsrc_v4[i]);
1666 if (priv->udpsrc_v6[i]) {
1667 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
1668 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
1669 gst_bin_remove (bin, priv->udpsrc_v6[i]);
1671 if (priv->udpsink[i])
1672 gst_bin_remove (bin, priv->udpsink[i]);
1673 if (priv->appsrc[i])
1674 gst_bin_remove (bin, priv->appsrc[i]);
1675 if (priv->appsink[i])
1676 gst_bin_remove (bin, priv->appsink[i]);
1677 if (priv->appqueue[i])
1678 gst_bin_remove (bin, priv->appqueue[i]);
1680 gst_bin_remove (bin, priv->tee[i]);
1681 if (priv->funnel[i])
1682 gst_bin_remove (bin, priv->funnel[i]);
1684 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1685 gst_object_unref (priv->recv_sink[i]);
1686 priv->recv_sink[i] = NULL;
1688 priv->udpsrc_v4[i] = NULL;
1689 priv->udpsrc_v6[i] = NULL;
1690 priv->udpsink[i] = NULL;
1691 priv->appsrc[i] = NULL;
1692 priv->appsink[i] = NULL;
1693 priv->appqueue[i] = NULL;
1694 priv->tee[i] = NULL;
1695 priv->funnel[i] = NULL;
1697 gst_object_unref (priv->send_src[0]);
1698 priv->send_src[0] = NULL;
1700 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1701 gst_object_unref (priv->send_src[1]);
1702 priv->send_src[1] = NULL;
1704 g_object_unref (priv->session);
1705 priv->session = NULL;
1707 gst_caps_unref (priv->caps);
1710 priv->is_joined = FALSE;
1711 g_mutex_unlock (&priv->lock);
1722 * gst_rtsp_stream_get_rtpinfo:
1723 * @stream: a #GstRTSPStream
1724 * @rtptime: result RTP timestamp
1725 * @seq: result RTP seqnum
1727 * Retrieve the current rtptime and seq. This is used to
1728 * construct a RTPInfo reply header.
1730 * Returns: %TRUE when rtptime and seq could be determined.
1733 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1734 guint * rtptime, guint * seq)
1736 GstRTSPStreamPrivate *priv;
1737 GObjectClass *payobjclass;
1739 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1740 g_return_val_if_fail (rtptime != NULL, FALSE);
1741 g_return_val_if_fail (seq != NULL, FALSE);
1743 priv = stream->priv;
1745 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1747 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1748 !g_object_class_find_property (payobjclass, "timestamp"))
1751 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1757 * gst_rtsp_stream_get_caps:
1758 * @stream: a #GstRTSPStream
1760 * Retrieve the current caps of @stream.
1762 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1766 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1768 GstRTSPStreamPrivate *priv;
1771 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1773 priv = stream->priv;
1775 g_mutex_lock (&priv->lock);
1776 if ((result = priv->caps))
1777 gst_caps_ref (result);
1778 g_mutex_unlock (&priv->lock);
1784 * gst_rtsp_stream_recv_rtp:
1785 * @stream: a #GstRTSPStream
1786 * @buffer: (transfer full): a #GstBuffer
1788 * Handle an RTP buffer for the stream. This method is usually called when a
1789 * message has been received from a client using the TCP transport.
1791 * This function takes ownership of @buffer.
1793 * Returns: a GstFlowReturn.
1796 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1798 GstRTSPStreamPrivate *priv;
1800 GstElement *element;
1802 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1803 priv = stream->priv;
1804 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1805 g_return_val_if_fail (priv->is_joined, FALSE);
1807 g_mutex_lock (&priv->lock);
1808 if (priv->appsrc[0])
1809 element = gst_object_ref (priv->appsrc[0]);
1812 g_mutex_unlock (&priv->lock);
1815 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1816 gst_object_unref (element);
1824 * gst_rtsp_stream_recv_rtcp:
1825 * @stream: a #GstRTSPStream
1826 * @buffer: (transfer full): a #GstBuffer
1828 * Handle an RTCP buffer for the stream. This method is usually called when a
1829 * message has been received from a client using the TCP transport.
1831 * This function takes ownership of @buffer.
1833 * Returns: a GstFlowReturn.
1836 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1838 GstRTSPStreamPrivate *priv;
1840 GstElement *element;
1842 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1843 priv = stream->priv;
1844 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1845 g_return_val_if_fail (priv->is_joined, FALSE);
1847 g_mutex_lock (&priv->lock);
1848 if (priv->appsrc[1])
1849 element = gst_object_ref (priv->appsrc[1]);
1852 g_mutex_unlock (&priv->lock);
1855 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1856 gst_object_unref (element);
1863 /* must be called with lock */
1865 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1868 GstRTSPStreamPrivate *priv = stream->priv;
1869 const GstRTSPTransport *tr;
1871 tr = gst_rtsp_stream_transport_get_transport (trans);
1873 switch (tr->lower_transport) {
1874 case GST_RTSP_LOWER_TRANS_UDP:
1875 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1881 dest = tr->destination;
1882 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1887 min = tr->client_port.min;
1888 max = tr->client_port.max;
1892 GST_INFO ("adding %s:%d-%d", dest, min, max);
1893 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1894 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1896 GST_INFO ("setting ttl-mc %d", ttl);
1897 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1898 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1900 priv->transports = g_list_prepend (priv->transports, trans);
1902 GST_INFO ("removing %s:%d-%d", dest, min, max);
1903 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1904 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1905 priv->transports = g_list_remove (priv->transports, trans);
1909 case GST_RTSP_LOWER_TRANS_TCP:
1911 GST_INFO ("adding TCP %s", tr->destination);
1912 priv->transports = g_list_prepend (priv->transports, trans);
1914 GST_INFO ("removing TCP %s", tr->destination);
1915 priv->transports = g_list_remove (priv->transports, trans);
1919 goto unknown_transport;
1926 GST_INFO ("Unknown transport %d", tr->lower_transport);
1933 * gst_rtsp_stream_add_transport:
1934 * @stream: a #GstRTSPStream
1935 * @trans: a #GstRTSPStreamTransport
1937 * Add the transport in @trans to @stream. The media of @stream will
1938 * then also be send to the values configured in @trans.
1940 * @stream must be joined to a bin.
1942 * @trans must contain a valid #GstRTSPTransport.
1944 * Returns: %TRUE if @trans was added
1947 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1948 GstRTSPStreamTransport * trans)
1950 GstRTSPStreamPrivate *priv;
1953 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1954 priv = stream->priv;
1955 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1956 g_return_val_if_fail (priv->is_joined, FALSE);
1958 g_mutex_lock (&priv->lock);
1959 res = update_transport (stream, trans, TRUE);
1960 g_mutex_unlock (&priv->lock);
1966 * gst_rtsp_stream_remove_transport:
1967 * @stream: a #GstRTSPStream
1968 * @trans: a #GstRTSPStreamTransport
1970 * Remove the transport in @trans from @stream. The media of @stream will
1971 * not be sent to the values configured in @trans.
1973 * @stream must be joined to a bin.
1975 * @trans must contain a valid #GstRTSPTransport.
1977 * Returns: %TRUE if @trans was removed
1980 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1981 GstRTSPStreamTransport * trans)
1983 GstRTSPStreamPrivate *priv;
1986 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1987 priv = stream->priv;
1988 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1989 g_return_val_if_fail (priv->is_joined, FALSE);
1991 g_mutex_lock (&priv->lock);
1992 res = update_transport (stream, trans, FALSE);
1993 g_mutex_unlock (&priv->lock);